For example, Japanese Patent Laid-Open Publication No. 9-152890 discloses, in the voice communication system, a method of, when a user desires low speed conversation, reducing the speaking speed of a received voice in accordance with the difference of the speaking speed between the received voice and a transmitted voice, whereby the received voice is made easy to hear.
FIG. 7 is a configuration diagram of a first prior art for realizing the above method. In FIG. 7, the speaking speed of a receiving signal and the speaking speed of a transmission signal, which is obtained by conversion of a transmitted voice through a microphone 702, are calculated respectively by speaking speed calculation parts 701 and 703.
A speed difference calculation part 704 detects a difference in speed between the speaking speeds calculated by the speaking speed calculation parts 701 and 703. A speaking speed conversion part 705 then converts the speaking speed of the receiving signal based on a control signal corresponding to the speed difference calculated by the speed difference calculation part 704 and outputs a signal, which is obtained by the conversion and serves as a received voice, from a speaker 706 including an amplifier.
When a predetermined receiving volume is used, a received voice is sometimes buried in ambient noise, and thus may be hard to hear. Therefore, in order to make the received voice easy to hear, a speaker should speak with a loud voice, or a hearer should manually adjust the receiving volume by, for example, turning up the volume. Thus, for example, Japanese Patent Laid-Open Publication No. 6-252987 discloses a method of automatically making a received voice easy to hear. In this method, the tendency that a hearer speaks generally louder when a received voice is hard to hear (Lombard effect) is used, and when a transmitted voice level is not less than a predetermined reference value, the receiving volume is increased, whereby the received voice is automatically made easy to hear.
FIG. 8 is a configuration diagram of a second prior art for realizing the above method. FIG. 8 is a configuration example of a voice communication system such that, a voice signal, which is transmitted and received with respect to a communication network 801 through a communication interface part 802, is input and output in a transmission part 805 and a receiving part 806. For example when the system is a cell phone, an overall control part 804 controls calling and so on based on key input information input from a key input part 803 for inputting a phone number and so on.
In FIG. 8, a transmitted voice level detection part 807 detects a transmitted voice level of a transmission signal output from the transmission part 805. Under the control of the overall control part 804, a received voice level management part 808 generates a control signal for controlling a received voice level based on the transmitted voice level detected by the transmitted voice level detection part 807.
A received voice amplifying part 809 controls an amplification degree of a received signal, which is received from the communication network 801 through the communication interface part 802, based on the control signal of the received voice level output from the received voice level management part 808.
The receiving part 806 then outputs a received voice from a speaker (not shown) based on the received signal with the controlled received voice level received from the received voice amplifying part 809.